Displaying 20 results from an estimated 121 matches for "list_route".
2009 Aug 04
4
Calling issue for non-extension numbers
...bc)/video=0x0 (nothing), combined - 0xc
(ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event),
peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event) Peer
audio RTP is at port 192.168.111.183:49152 Looking for 6789940793 in
DLPN_Free_Outbound (domain sip.speartek.com) list_route: hop:
<sip:103 at 192.168.111.183>
It appears that my device cannot connect to the server when dialing
certain numbers. Does anyone have any idea about this?
Thanks,
Kayton
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2005 Feb 26
0
NAT= setting for a public proxy
...nd description format pcma
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x10f
(g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1
(g723)
Looking for 10 in extensions
list_route: hop: <sip:abpusa.com:5060;maddr=209.189.239.106;lr=1>
list_route: hop:
<sip:209.189.239.106:5062;transport=udp;dest=4.13.144.17-5060;to-tag=u2dogw3pvf;lr=1>
list_route: hop: <sip:9723048721@192.168.5.102:5060;line=70n796w7>
Feb 26 09:10:51 NOTICE[10677]: chan_sip.c:3084 copy_via...
2005 Sep 14
2
Starting From Scratch
...Peer audio RTP is at port 66.81.0.97:16488
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0
(nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined -
0x1 (g723)
list_route: hop: <sip:916xxx6000@66.81.0.87;r2=on;ftag=as0dbb6283;lr=on>
list_route: hop: <sip:916xxx6000@192.168.4.16;r2=on;ftag=as0dbb6283;lr=on>
list_route: hop: <sip:916xxx6000@192.168.4.97>
set_destination: Parsing
<sip:916xxx6000@66.81.0.87;r2=on;ftag=as0dbb6283;lr=on> for
addres...
2005 May 13
0
asterisk dials random number when receiving incoming call
...one-event
Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0xe
(gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Looking for 448715046363 in local-sip
list_route: hop: <sip:82.70.154.145;ftag=as3606b893;lr=on>
list_route: hop: <sip:Unavailable@213.166.5.129>
Transmitting (NAT) to 82.70.154.145:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
82.70.154.145;branch=z9hG4bK2922.6c170001.0;received=82.70.154.145;rport=5060
Via: SIP/2.0/UDP 213.166.5.129:506...
2005 Jan 07
0
Inbound Pickup Issue - Sipmedia
...RTP is at port 209.247.23.201:60062
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing)
Found peer 'Sipmedia'
Looking for s in from-Sipmedia
list_route: hop: <sip:+1Myphonenumber@69.1.236.33;r2=on;ftag=VPSF50603522629637;lr>
list_route: hop: <sip:+1Myphonenumber@69.1.236.33;transport=tcp;r2=on;ftag=VPSF50603522629637;lr>
list_route: hop: <sip:209.247.16.5:5060;transport=tcp>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/...
2003 Dec 20
2
More beginner questions
...-16
12 headers, 10 lines
Found audio format UNKN
Found audio format UNKN
Found audio format UNKN
Found description format GSM
Found description format PCMU
Found description format telephone-event
Capabilities: us - 6, them - 6/0, combined - 6
Non-codec capabilities: us - 1, them - 1, combined - 1
list_route: hop: <sip:612@192.246.69.223;ftag=as1f0e4544;lr=on>
list_route: hop: <sip:30342@65.121.72.14>
set_destination: Parsing <sip:612@192.246.69.223;ftag=as1f0e4544;lr=on> for
addr
ess/port to send to
set_destination: set destination to 192.246.69.223, port 5060
Transmitting:
ACK sip:3...
2005 Jan 09
0
RE: Inbound calls getting disconnected when I answer the phone, using 'SIP'
...RTP is at port 209.247.23.201:60062
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing)
Found peer 'Sipmedia'
Looking for s in from-Sipmedia
list_route: hop: <sip:+1Myphonenumber@69.1.236.33;r2=on;ftag=VPSF50603522629637;lr>
list_route: hop: <sip:+1Myphonenumber@69.1.236.33;transport=tcp;r2=on;ftag=VPSF50603522629637;lr>
list_route: hop: <sip:209.247.16.5:5060;transport=tcp>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/...
2003 Oct 03
4
Iconnect Incomming calls
...6 RTP/AVP 4 18 101 19
c=IN IP4 213.137.65.234
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
23 headers, 14 lines
Ignoring this request
Looking for 14103445557 in sipinbound
RDNIS is 4103445557
list_route: hop: <sip:14103445557@213.137.73.178:5060;maddr=213.137.73.176>
list_route: hop: <sip:4103532264@213.137.65.234:5060>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 213.137.73.176:5060;branch=6acca267-67952546-2efdf97b-e80aa0ef-1
Via: SIP/2.0/UDP 213.137.65.234:5060
From:...
2013 Dec 31
2
*8 and SIP
...w|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined -
0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 172.16.10.101:11792
Looking for *8 in trunk_office (domain 10.65.6.10)
list_route: hop: <sip:nicktest at 172.16.10.101:5063>
<--- Transmitting (NAT) to 208.65.55.170:5063 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
172.16.10.101:5063;branch=z9hG4bK908225576;received=208.65.55.170
From: "nicktest" <sip:nicktest at 10.65.6.10>;tag=1470823868
To: <...
2005 Jan 09
1
Inbound calls getting disconnected when I answer the phone, using 'SIP'.
...RTP is at port 209.247.23.201:60062
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined - 0x0 (nothing)
Found peer 'Sipmedia'
Looking for s in from-Sipmedia
list_route: hop: <sip:+1Myphonenumber@69.1.236.33;r2=on;ftag=VPSF50603522629637;lr>
list_route: hop: <sip:+1Myphonenumber@69.1.236.33;transport=tcp;r2=on;ftag=VPSF50603522629637;lr>
list_route: hop: <sip:209.247.16.5:5060;transport=tcp>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/...
2004 Aug 26
0
Asterisk media problem behind NAT
...mat UNKN
Found audio format ALAW
Found audio format UNKN
Found audio format UNKN
Found video format UNKN
Found video format UNKN
Found video format UNKN
Capabilities: us - 786446, them - 303/851968, combined - 786446
Non-codec capabilities: us - 1, them - 0, combined - 0
Looking for 3004 in default
list_route: hop: <sip:<gateway1>:5060;transport=udp>
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.1.54:5060;branch=z9hG4bK3f113fc0c05ec1deece622bd0ed4a521;received=<gateway1>
From: "3002" <sip:3002@<asterisk ip>:5060;transport=udp>;tag=24957277
To: <...
2003 Aug 21
0
No audio in either direction, sip channels hanging, asterisk will not shut down.
...000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,16
14 headers, 9 lines
Found audio format 0
Found audio format 101
Found description format PCMU
Found description format telephone-event
Capabilities: us - 524302, them - 4/0, combined - 4
Non-codec capabilities: us - 1, them - 1, combined - 1
list_route: hop: <sip:62.254.245.14:5060;lr=1>
list_route: hop: <sip:3046@vega.culver-tec.com:5060;maddr=62.254.245.12>
set_destination: Parsing <sip:62.254.245.14:5060;lr=1> for address/port to send to
set_destination: set destination to 62.254.245.14, port 5060
Transmitting:
ACK sip:3046@s...
2007 Apr 11
1
Mediatrix 1204
...scription format PCMA
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10d
(g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1
(g723)
Looking for 4000 in from-pstn
list_route: hop: <sip:3330001@192.168.0.253>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK2a550f4da
From: Incoming <sip:3330001@192.168.0.254>;tag=68b2ce27259fa46
To: sip:4000@192.168.0.254
Call-ID: 9dd4c369dbcf118b22b91aa08728f726@192.168.0.254
CSeq: 5271...
2003 Jul 01
0
"Forbidden" problem!!
...a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
11 headers, 13 lines
Using latest request as basis request
Sending to 10.8.210.147 : 5060 (non-NAT)
Capabilities: us - 12, them - 526, combined - 12
Non-codec capabilities: us - 1, them - 1, combined - 1
Looking for 00115601992 in dialout
list_route: hop: <sip:g.carnero@10.8.210.147:5060>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.8.210.147:5060
From: Giorgio <sip:g.carnero@asterisk>;tag=659649795
To: <sip:00115601992@asterisk>;tag=as15fbf9fc
Call-ID: 8A229D4B-4B34-484D-B370-E9DF6867F411@10.8.210.147...
2008 Apr 03
0
Problems with analog <-> SIP phone confif\gurations
...Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x4
(ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:Ananth2 at 192.168.102.10>
set_destination: Parsing <sip:Ananth2 at 192.168.102.10> for address/port
to send to
set_destination: set destination to 192.168.102.10, port 5060
Transmitting (NAT) to 192.168.209.1:48099:
ACK sip:Ananth2 at 192.168.102.10 SIP/2.0
Via: SIP/2.0/UDP...
2008 Apr 04
0
Problems with Analog - SIP phone conversations
...Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x4
(ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:Ananth2 at 192.168.102.10 <sip%3AAnanth2 at 192.168.102.10>>
set_destination: Parsing
<sip:Ananth2 at 192.168.102.10<sip%3AAnanth2 at 192.168.102.10>>
for address/port
to send to
set_destination: set destination to 192.168.102.10, port 5060
Transmitting (NAT) to 1...
2005 Sep 29
1
Cisco AS5300 --> [SIP] --> Asterisk - NO AUDIO
...ion format X-NSE
Capabilities: us - 0x100 (g729), peer - audio=0x30f
(g723|gsm|ulaw|alaw|g729|speex)/video=0x0 (nothing), combined - 0x100
(g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0
(nothing), combined - 0x0 (nothing)
Looking for 84104214 in default (domain 70.84.200.204)
list_route: hop: <sip:203.88.192.42:5160;ftag=1CA65AC-9C8;lr=on>
Transmitting (no NAT) to 203.88.192.42:5160:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
203.88.192.42:5160;branch=z9hG4bK5aeb.8f7e6027.0;received=203.88.192.42
Via: SIP/2.0/UDP 211.147.240.237:5060;rport=57786
From: <sip:0017911@211.147.240.2...
2004 May 28
2
Asterisk with Draytek 2600V
...rmat telephone-event
Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer -
audio=0x11d(G723|ULAW|ALAW|G726|G729A)/video=0x0(EMPTY), combined -
0xc(ULAW|ALAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined -
0x1(G723)
Found user 'phone1'
Looking for 90800500005 in sip
list_route: hop: <sip:phone1@192.168.1.1>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-Ifn-9746
From: phone1 <sip:phone1@192.168.0.250:5060>;tag=eSJ-4736
To: <sip:90800500005@192.168.0.250>;tag=as71701551
Call-ID: diY-24872@192.168.1.1
CSeq: 1 IN...
2012 Dec 10
1
Problem with SIP trunk I've set up between two * boxes.
...), peer - audio=0xe
(gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 172.17.9.1:4006
Looking for 7444 in local_SIP (domain 172.17.0.17)
list_route: hop: <sip:6110 at 172.17.9.1:55388;ob>
<--- Transmitting (no NAT) to 172.17.9.1:55388 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
172.17.9.1:55388;branch=z9hG4bKPjvX4It3-WYRqMlyhU9peo5ewQRIgQ4qd1;received=172.17.9.1;rport=55388
From: <sip:6110 at 172.17.0.17>;tag=UzpwMu8tx77dICqA...
2005 Jan 14
1
ULaw not negotiating
...on format PCMU
Found description format telephone-event
Capabilities: us - 0x4(ULAW), peer - audio=0x4(ULAW)/video=0x0(EMPTY),
combined - 0x4(ULAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined -
0x1(G723)
Found peer 'broadvoice'
Looking for 5555551212 in incoming
list_route: hop:
<sip:5555551313@147.135.4.128:5060;bvoice=ACME-asdfadf2dfsa3;ep=147.135.4.12
9;transport=udp>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
147.135.4.128:5060;branch=jkjk245kjelkjelkj2435sadflkj435.1sr
From: "TEST
PHONE"<sip:5555551313@147.135.4.129;user=ph...